® 2N NETSTAR Communication switch Configuration tool Version 2.6.0 www.2n.
The 2N TELEKOMUNIKACE joint-stock company is a Czech manufacturer and supplier of telecommunications equipment. The product family developed by 2N TELEKOMUNIKACE a.s. includes GSM gateways, private branch exchanges (PBX), and door and lift communicators. 2N TELEKOMUNIKACE a.s. has been ranked among the Czech top companies for years and represented a symbol of stability and prosperity on the telecommunications market for almost two decades.
Table of Content 1 Basic Information ............................................................................................................ 5 1.1 About Help ..................................................................................................................................................................5 1.2 About application.........................................................................................................................................................5 1.
8 Users ............................................................................................................................. 98 8.1 Users and Groups ....................................................................................................................................................... 98 8.2 User rights ............................................................................................................................................................... 104 8.3 Station types .
About Help 1.1 1 Basic Information 1.1 About Help The document serves as Help and Manual for the configuration of the communication system 2N Netstar by program NsAdmin. 2N reserves the right to modifications. 1.2 About application About application NsAdmin is a configuration tool that is used to configure 2N Netstar communication system, version 2. The application is designed for a x86 platform using the WINDOWS 2000/XP operating system connected within a network with 2N Netstar.
Connection to the PbX 1.3 Size of indent – It defines size of indent for xml trace. Colours Setting colours of ports – Here you can define colour of each type of the virtual port or completely disable this function. Setting colours of tabs – Here you can define background color for tabs or completely disable this function. Setting colours of stations – Here you can define colour of each type of the station (SIP, external, ...) or completely disable this function.
Connection to the PbX 1.3 Cancel auto login – With this option you can cancel automatic connection. You haven't to select concrete object before. Except these options are within context menu another options with following meaning: Import PbX structure – With this option you can import predefined PbX structure, which is described below. Export PbX structure – With this option you can export actual PbX structure, which you can later use for work from another computer.
Connection to the PbX 1.3 Delete autosave item older – This parametr sets maximum time for keeping old database backups in the storage folder. This function can be used only for offline mode. Autosynchronization – This option enables automatic synchronization between offline database and database in the PbX. This function can be used only for offline mode. If it is active then these modes are used : Loading from PbX if the offline database is empty.
Connection to the PbX 1.3 Figure 4 View of the properties setting of the connection. Connecting to the PbX After automatic or manual initiation of connecting to the selected PbX is displayed dialogue from figure 5. In this dialogue you can find information about connected PbX and also information about version of firmware of the PbX (if detected). You can also see information about last knoen error of connection and in the case of automatic connection attemps also remaining time to another attempt.
Configuration menu 1.4 Figure 5 View of the dialogue of connection course. If you aren't able to connect to your PbX, please check following: 1) PbX is switched on; 2) PbX is connected to the network; 3) both sides have the same IP address and port; 4) used communication port is opened; 5) you are using corresponding versions of firmware and configuration tool; 6) used communication port isn't blocked by your antivirus software; 1.
Configuration menu 1.4 Exit – It is used for exiting configuration tool. Trace Load trace from file – This option is used for loading concrete saved trace from the file. Old trace is cleared. Add trace from file – This option is used for adding another saved trace to actual one. It can connect more traces to one, which you can analyse easily. Save trace to file – This option is used for saving actual online trace to the file.
Configuration menu 1.4 Figure 2 View of dividing configuration tool to windows and menu tabs. Important parts of the configuration tool are also menu tabs Trace and Database, which you can find on the bottom of the configuration tool above the status bar. Within the menu tab Trace you can trace call signalization and communication at all interfaces of the PbX. With this trace you can easily detect main problems and mistakes in your configuration.
PbX activation 1.5 1.5 PbX activation What do you need? To activate and set up 2N Netstar you need the 2N Netstar itself, x86 computer running supported operating system Windows, keyboard and mouse. PbX have to be connected in the network with 2N Netstar. It is also necessary to display the redirected standard input and output of 2N Netstar on your PC's console. For this you need a six-core cable with a six-pin connector RJ-12 on one end and a serial connector on the other end.
PbX activation 1.5 Figure 1 View of dialog box of wizard for hardware configuration. If wizard was displayed, in this step you are able to define basic configuration of BRI virtual ports. This setting can be changed at any time and if you aren't sure how to set it, you can go to another step via button Next. After it begins configuration tool (in cooperation with PbX) to detect used hardware, as you can see on figure 2. Figure 2 View of the wizard during hardware detection.
PbX activation 1.5 Detected aren't only boards in the basic case, but also in connected extenderes. As soon as is hardware detection finished, are created virtual ports for all detected ports (except VoIP boards). After it will be your PbX ready for other configuration. Active hardware is signallized by LED diods. Each board has to light green, except GSM board which doesn't have any LED diod for signalling of board state. However functional GSM board id indicated by diods of its ports.
PbX activation 1.5 Figure 4 View of the wizard for creating stations or for import of company structure. Step 5: Creation of the routers Last step of this wizard is used for routers creation. Routers are objects used for call or SMS routing through the PbX from one port to another. Wizard offers some default sets of routers, which are sufficient for basic call routing.
Hardware profiles 2.1 2 Hardware 2.1 Hardware profiles Setting up of hardware profiles you can find in the menu HW – HW profiles. With this setting you can use your system more efficiently in some specific cases. This menu contains five different hardware profiles. Its benefites and disadvantages are evident from following table. Table 1 Shows benefits and disadvantages of each of five hardware profiles. Enabled – With this option you can disable or enable using of this rack.
Boards 2.2 After clicking on the right mouse button within the view of boards are available following options: Add board – This option can be used only when context menu was invoked on free position of the case (without any board). You can add board which is detected or your own from the list of supported boards for this possition. Remove board – This option is used for removing selected board. If this board has assigned virtual ports, you can remove them also or retain.
Boards 2.2 Figure 2 View of analog board with all possible states of signalization. Cross Green – Physic port with assigned virtual port. Yellow – Physic port with assigned virtual port and active call (or call establishment). Earphone Green – Physic port with assigned virtual port and assigned station. Yellow – Physic port with assigned virtual port, assigned station and active call (or call establishment).
Synchronization 2.3 Tab Virtual port Tab Virtual port is used for easier setting up of virtual ports. In this tab you can set up all parameters of virtual ports and simultaneously see displayed screen of case. Setting of parameters of virtual ports is described in chapter 3 according to type of stack. Addressing The position of each board is specified in the format R : C : B and the position of a port in the format R : C : B : P.
Board list 2.4 2.4 Board list This overview you can find in the menu Hardware – Board list. Within this menu is displayed list of boards, which are physically present in the PbX. Boards list is displayed in four columns with following meaning: Address – It shows physical board address within the PbX according to the chapter Addressing. Type – It shows board type. Serial number – It shows board serial number, which was burned at production. MAC address – It shows board MAC address. 2.
BRI and PRI virtual ports 3.1 3 Virtual ports 3.1 BRI and PRI virtual ports 3.1.1 BRI virtual port BRI carriers are assigned to physic ports of ISDN boards for Basic Rate Interface. Hardware configuration of BRI carriers you can find in the menu Virtual ports – BRI/PRI on the tab Stack. On the left side is displayed list of all BRI carriers and on the right side of this menu you can set up parameters of selected carrier. Whole configuration is divided into logical parts.
BRI and PRI virtual ports 3.1 line with higher bit error rate must BER rate go under lower level. Interval between these two values is used as hysteresis. BEr values are entered in exponential form (e.g. 3e-5 means 3 errors in 100000 bits). Figure 1 View of jumper configuration for each type of ISDN BRI board. Thick line presents front side of the board. Specific interface parameters Multiframe – It is parameter of first layer for So bus. Further information you can find in recommendation I.430.
BRI and PRI virtual ports 3.1 Terminals – This field is activ only for virtual ports on mode NT with bus mode MPT and you have to enter there all connected terminals and theirs MSN numbers. To these terminals you can after it assign stations. Terminal with concrete MSN number then uses identification of assigned station. Digital interface diagnostic Line state – Parameter can't be set. It shows only state of the first layer of the interface.
BRI and PRI virtual ports 3.1 Settings for SLIP – With parameter Nonsynchronous as error you can enable another parameters for setting SLIP range. This option can be used only on TE port. If SLIP rate gets over upper level, is this fact signalized by red exclamation on the port within menu Hardware – Boards and by red text in the field for stact status. For changing status of nonsynchronous line must SLIP rate go under lower level. Interval between these two values is used as hysteresis.
Cornet virtual port 3.2 layer. Caution is made by red exclamation on the port within menu Hardware – Boards and by red text in the field for stact status. This option can't be combined with option Disconnect L2. Digital interface diagnostic Line state – Parameter can't be set. It shows only state of the first layer of the interface. Number of SLIPs per minute – This parameter shows number of SLIPs. SLIP is caused by different clock on devices on this interface (PbX and terminal).
ASL virtual port 3.3 error status is changed after BER value falls below BER OK level. Range between values is used as hysterezis. BER value is entered in ecponential form (e.g. 3e-5 means 3 errors in 100000 bits). Master terminal Type – Shows type of connected terminal StarPoint. Firmware – Shows used firmware of connected terminal. Extenders – Shown information about used extenders of connected terminal. Slave terminal Type – Shows type of connected terminal StarPoint.
ASL virtual port 3.3 Figure 1 View of hardware configuration for ASL virtual port. Line parameters Impedance – This parameter determines impedance of the hybrid according to preset models (User, ETSI 600, Germany and Real 600). Line model – This parameter determines another parameters of the hybrid according to preset models EIA0 to EIA7 (e.g. EIA0 represents model of 100m long line). Signalling type – Type of signalling of Active state. You can choose between Reverse polarity, Tarif pulse or Simple.
CO virtual port 3.4 3.4 CO virtual port This is the analog carrier for connection of state analog line (central exchange). This carrier dispose only with DTMF transmitter and that is why you have to route incoming call directly to final destination or into DISA object. With DISA you are able to detect DTMF symbols and route the call to specific destination. Parametres are divided into logical sections. Stack status This field displays information about stack and its actual state.
GSM virtual port 3.5 Tariff pulse type – It defines source for tariff pulse sending. You can choose between 12 kHz, 16 kHz or none. Outbound way parameters (from PbX) Current detection timeout [ms] – Time for detection of the current on the picked-up carrier. If no current is detected in this time, a failure is reported. Dial tone wait timeout [ms] – This parameter specifies the time of waiting prior to dialing the numbers to the carrier.
GSM virtual port 3.5 Stack status This field displays information about stack and its actual state. Network selection Roaming enabled – With this option you can enable roaming for concrete GSM virtual port. Manual network selection – If this option isn't checked, inserted SIM tries to log into the prefered network automatically. If it is checked, you have to enter correct network code and inserted SIM card tries to log only into this concrete network.
GSM virtual port 3.5 Figure 1 View of hardware configuration for GSM virtual port. GSM interface parameters Hide number – This checkbox enables calling with suppressed calling party number (CLIR – Calling Line Identification Restriction). This function must be supported by the network (by the operator). Otherwise calls with suppressed identification are refused with corresponding cause. Relax timeout between calls – This parameter determines the idle period between two calls.
SIP virtual ports 3.6 Module IMEI – This field shows detected IMEI number. GSM network diagnostic State – This field shows actual state of the port. According to this state you are able to detect some problems with logging into the network. For example the state PIN REQUESTED means, that SIM card needs PIN code, which you have to insert or you can also disable PIN requesting at the SIM card. Otherwise the SIM card can't be logged into the network. Logged network – This field shows actual network code.
SIP virtual ports 3.6 Figure 1 View of configuration menu for SIP gateway. Connect to gateway Host – It defines Realm of the other side of trunk connection (VoIP provider or another PbX). Port – It defines port on the other side of trunk connection. To this port you have to send SIP messages. Register line – It enables line registration. If line isn't registered, no call establishments are done on this virtual port. All attempts are rejected with corresponding cause within PbX.
SIP virtual ports 3.6 Codecs Supported – In this field you can find list of supported codecs. You can't find here codecs which were selected as used. Selected – In this field you can find list of codecs, which are used for communication on this virtual port. Within context menu is another option for setting features of selected codec. DTMF according to RFC2833 – With this option you can enable transmission of DTMF according to RFC2833. Fax T.
SIP virtual ports 3.6 assigned to the specific terminal on the tab Stations. Figure 3 View of configuration part related only to SIP proxy. Proxy parameters Registrations validity – With this parameter you can define time of validity for terminal registrations. After expiration each terminal has to send new register request. Parameter could be set in range 30 – 3600s. Terminals This section is used for terminals management. If terminals aren't created then VoIP phones can't register to this SIP proxy.
SIP virtual ports 3.6 port. Within context menu is another option for setting features of selected codec. DTMF according to RFC2833 – With this option you can enable transmission of DTMF according to RFC2833. Fax T.38 – With this option you can enable fax transmission according to recommendation T.38. If this option is checked, it is changed to link with advanced settings. Recommended setting is TCF – Transfer, Error correction – Redundancy and No compression. 3.6.
Virtual port options 3.7 DSP – This section can be used for transferred data optimalisation. When it is enabled packets aren't sent needlessly if user doesn't speek. Shortcut VAD means Voice Activity Detector. Generate comfort-noise – With this parameter you can enable generating of some noise to the background of the call. Mask lost packets – With this option you can activate optimalisation for lost packets masking. It can be used only by lower error rate, because it is based on some presumption.
Virtual port options 3.7 tabs are used only at few virtual port types, because on others they haven't any sense. Basic The tab Basic brings following parameters: Name according to the physic port – With this option you can change name of the virtual port according to the physic port where it is assigned. This name consists of stack name and of hardware address in the square brackets. In the case of manual change of this name is option automatically unchecked.
Virtual port options 3.7 not. Autoclip routers section This part is used for assigning specific autoclip router to the virtual port. You have to assign autoclip router for calls and for messages separately, but you can assign the same autoclip router to both of them. More information about autoclip routers you can find in the chapter 7.4 Autoclip routers. Calls – Here you can assign autoclip router for saving records of outgoing calls, which weren't accepted or were rejected by opposite user.
Virtual port options 3.7 that public network. Disconnect tone is mostly generated by own PbX. Overlap Overlap is one of two methods of called party number (CPN) sending. In the case of overlap dialing isn't called party number transmitted all in the SETUP message, but it is transmitted digit by digit in the INFO message. Setup consists of following parameters: Overlap sending – This parameter enables overlap sending in the direction from port into the PbX. Above all it is used at ISDN virtual ports.
Virtual port options 3.7 Stack The tab Stack is described in the chapter 3 Virtual ports, because it depends on stack type. Messages The tab Messages is used only on the virtual port types, GSM and SIP virtual ports, because on the others it has no sense. This tab consists of following parameters: Repeat at fail – With this parameter you can enable or disable repeated attempts for transmitting received SMS messages.
SIM cards 4.1 4 SIM cards 4.1 SIM cards In the menu Carriers – GSM – SIM you can find list of all SIM cards of the PbX. After first inserting of the SIM card into the port is automatically created new record in the database and parameters which you fill (e.g. PIN), are used automatically by future detection of this SIM card. This menu includes following parameters: SimNum – This parameter shows detected identification number of the SIM card.
Network interface 5.1 5 Network 5.1 Network interface In the menu Network – Network interface you can manage all network interfaces, which are present in the PbX. In addition to CPU interface are there ethernet interfaces of Surf boards. Bit rare of all interfaces is 100 Mbit/s. This interfaces are used for communication with PbX, SMTP clients and for signalling and RTP streams of VoIP calls.
Services Setting 5.2 Figure 1 View of configuration of SMTP clients. Mail account – It shows only name of selected SMTP client. It can't be directly modified. Network interface – Here you can choose network interface used for SMTP communication with server. In this version you can choose for this purpose only network interface of CPU board. Outgoing mail server – Here you have to enter IP address of SMTP server. If you use DNS server you can use also domain name of your SMTP server.
Services Setting 5.2 Figure 1 View of menu for time synchronization. 5.2.3 Remote control (SNMP) What is it SNMP? The Simple Network Management Protocol (SNMP) forms part of the internet protocol suite as defined by the Internet Engineering Task Force (IETF). SNMP is used in network management systems to monitor network-attached devices for conditions that warrant administrative attention.
Services Setting 5.2 Password – Here you have to enter password for encryption. Access – Within this parameter you have to set rights of this user. You can do it by choosing from the list of created rights. Rights Add default rights – With this option of context menu you are able to create default rights Internet and Restricted. With these rights are created also filters on the tab Filters. Title – It defines name of created right. This title is displayed by rights choosing on the tab Users.
Services Setting 5.2 MIB files Add – With this option you can add MIB files used MIB database extension. Delete – With this option you can delete selected MIB file. Recompile – With this option you can recompile selected MIB file. File – In this column is shown path to the source of MIB file. This path is important for using option Recompile. State – This column shows actual state of MIB file. You can see here states Compiled, Not compiled and Not found.
Services Setting 5.2 Figure 3 View of the part for setting of listening port. Notification Client address – It defines IP address or domain name of client, where are sent notifications according to the below specified filter. Client port – It defines client port where are sent notifications. Used local port – If you need you can also specified port of the PbX which is used for sending notifications. Notification type – Within this section you can select type of used notification.
Services Setting 5.2 Figure 4 View of the configuration menu on the tab Notification. User/Community – Here you have to define concrete SNMP user, which corresponds to USM for SNMP v3 and to community for other versions. Filter – Here you have to define filter for notifications. Longer OID for root or subtrees means stricter filter. Security level – This parameter can be used only for SNMP v3 and it defines level of security for notifications.
Services Setting 5.2 Figure 1 View of configuration in menu Message-bridge. If you are connected to the PbX via port with authentication, basically you don't see tab Database, which can be used for direct configuration. If you want to see it, you have to assign rights for reading and writing for this user. It could be done within the menu Users – Users rights. Another possibility is to connect to your PbX via port without authentication. In such a case you expose your PbX to danger of unauthorized access.
Global parameters 6.1 6 Global data 6.1 Global parameters Switch on regime ME With this option you can switch your PbX to regime Mobility Extension, which is used in special cases when is PbX connected as gateway between another PbX and different types of private or public networks. When this regime is activ, all Flash patterns and DTMF characters are sent directly to the specified port. PbX in this mode does't react to these patterns and DTMF characters.
Localization 6.2 concrete prefix to virtual port via parameter Added prefix for external CLIP, which you can find on the tab Basic. Meaning of columns from tab is following: Prefix name – It defines name of prefix, which is used in another menus within configuration tool. Prefix – Here you have to define concrete prefix, which is then added to calling party identification. Group of users – In this column you are able to define groups of users, which can use this concrete prefix.
Licences 6.3 Local settings With checkbox Local calls possible you can open part of this menu, where you can setup national parameters similarly as in the part International: Number – This number presents national access code (area code). For example in Slovakia has city Bratislava area code 2. Prefixes – This prefix presents access codes for access to national telephone network.
Language packages 6.4 Licences This part displays transparent table with content of selected licence file. Field consists of some columns with following meaning: Licence type – It shows specific type of licensed service, interface or object within the system. Subtype – This column defines specific licence within its type. Owned – This column shows number of licensed channels, terminals or services.
Services 6.5 Figure 1 View of the menu where you can add language packages. Meaning of each column is following: Name – This column shows name of installed language package. Default packages are named according to their language. Storage – This column defines part of way to the package storage within the system data space. Builtin means directory /opt/netstar and Internal means directory /data/netstar. Together with column Directory gives absolute path to the storage.
Services 6.5 If you want to use some service, you have to dial specific prefix, which is routed to the routing table according to called number. Default router for services is situated in the menu Routing - Routers and is called Services: Figure 1 View of part of router Services, which is used for routing calls to services. Another possibility how to activate or deactivate some service is activation via messages.
Progress tones 6.6 6.6 Progress tones Introduction Progress tone is general name for all tones and announcements played within the PbX. After creating new database has PbX set of default progress tones. Amount of progress tones and language alternations depends on installed language packages. This basic set could be extended with own files, tones or by connecting of external audio inputs (e.g. mp3 player). Whole menu is logicaly divided into couple of tabs.
Progress tones 6.6 Play – It is used for playing selected progress. Progress configuration Action – In this column you can choose one of listed commands, which consequently defines meaning of that row. Repeat – It defines number of repeating. This command can be used only once within each progress and it means that all rows from the beginning to this row will be repeated. Number of repeating is set within the column Repeating.
Progress tones 6.6 Own files Own files This section shows all files uploaded by user to the PbX, which can be used as sources for progress. Context menu of this section has following functions: Add – Is used for adding record. This record is then used as source for progress. After creating it has no file. It has to be uploaded via section Own files source. Rename – With this function you can rename selected record. Delete – With this function you can delete selected record.
Progress tones 6.6 some of them were deleted. Made changes of other tones are preserved. Restore default tones – With this option you are able to complete list of default tones. All changes made within default tones are lost. Tone configuration In this section you can configure selected tone via table with three columns with following meaning. Language – This column defines language for each row. You are able to create different forms of tone for different languages.
Ring tones 6.7 Other sections Sections Information about progress and Progress configuration are common for all tabs and meaning of theirs parameters is described in part Progress list. 6.7 Ring tones Ring tones you can set up in the menu Tones & Rings - Ring patterns. Each ring tone consists of ring pattern and cornet melody. Some terminals aren't able to change ring melody and use only ring pattern. On the left side of the menu is displayed list of the created ring patterns.
Time parameters 6.8 ring pattern is then used with selected melody. Signal BRI – This parameter sets ring tone signaling for the ISDN terminals. This function has to be supported by used ISDN terminal or it will be ignored . Durations – Here you can set ring pattern via two columns and four rows. For each row which you want to use, you have to set column ON and column OFF. First one represents time of the ring current and second one represents timeout before resuming on the other row.
Time parameters 6.8 Figure 2 View of the dialogue box for setting date and time of the PbX. 6.8.2 Time conditions Time conditions you can set in the menu Global data – Time parameters – Time conditions. This menu is divided into two parts. On the left side is displayed list of defined time conditions, which you can create, remove or rename via context menu. On the right side you are able to set up selected time conditions. One time condition could consists of several simple rules, which are summed.
Time parameters 6.8 Figure 2 Part of the menu for time conditions editing. Regarding to complexity of the time conditions are defined exact rules: 1) There are defined parameters which optionally defines absolute limits of time interval validity (it means beginning and ending). If these limits are defined, time condition can be valid only within this limits, regardless further settings (including Interval negation). On the top of dialogue box for time interval setting are checkboxes From and To.
Autoclip parameters 6.
Assistant 6.10 Autoclip parameters Into the autoclip table you can save outgoing calls (outgoing messages) records with parameters dependent on the calling (sending) user. To this purpose serve just autoclip parameters. You can define it in the menu Global data – Autoclip parameters. This menu is divided into two parts. On the left side is displayed list of created sets of autoclip parameters. There you can add, remove or rename sets of autoclip parameters via context menu.
Assistant 6.10 history. If this option is checked, user is asked for confirmation before each record removing. Default language – With this option you can set up predefined application language. At this time you can choose between three languages – Czech, English and Finnish. Image directory – With this option you can set up using of one of predefined image sets. CSS style file name – With this option you can set up CSS style, which will be used within the application. Max.
Routers 7.1 7 Routing 7.1 Routers Router Router is the set of rules, which are used for incoming call routing through the PbX. Routers are defined in the menu Routing – Routers which consists of two windows. In the window on the left side is displayed list of created routers. On the right side you can set up selected router. On the left side of the menu you can use context menu with following options: Add – With this option you can initiate dialog box for adding routers.
Routers 7.1 format. Show objects routing into the router – With this option you can open side window where you can see list of all objects which are routed into selected router. Call routing Call routing is executed similarly within all router types. At first is found row according to incoming informations (called or calling number, called or calling number subtype, call type, incoming carrier or text of incoming SMS message) and then is used rule from that row.
Routers 7.1 Number – It means digits, letters A, B, C, D and characters *, #, +. , – Comma means waiting for one second. p(X) – Symbol X you have to replace by number of seconds you want to wait. This instruction corresponds to "X" commas. t – Parameter "t" sets if preset number is dialled after connect to the voice channel (parameter "t" is used) or if it is only dialled with delay before connect (parameter "t" isn't used). Scheme – In this column you can change called number scheme to number or URI.
Routers 7.1 columns with the same meaning as in the case of router by called number. Difference is only that in this case presents prefix calling party number and you can't use instructions for delayed dialling. Calling party number changes made within this router are used only for routing and not for call identification. c) By called number subtype This router is based on routing according to the called number subtype (CDN subtype).
Routers 7.1 and you can't change it. Calling party number subtype changes made within this router are used only for routing and not for call identification. e) By call type This router is based on routing according to the call type (voice, data, video, ...). All columns have the same meaning as in the case of router by called number subtype except first one. First column sets up call type. When is preset call type recognized, call is routed to the preset destination.
Routing objects 7.2 rule within row with time condition is valid only in the time when is valid preset time condition. Thanks time conditions you are able to create sophisticated routing according to the time. For the same incoming conditions (except time) you can route the call to different destinations.
Routing objects 7.2 Figure 1 View of configuration menu of routing object Bundle – tab Basic. This menu consists of many parameters with following meaning: Allocate strategy – This parameter sets way of object choosing within the selected bundle. You can choose between strategy Linear or Cyclic. Linear strategy – Incoming call is always routed to the first row of the bundle. If this object is busy or unavailable, call is routed to the next row or terminated (according to another setting).
Routing objects 7.2 incoming call isn't terminated after routing to the last unused object, but it rings to the last destination. Last unused object needn't to be the object from the last row of the bundle. Default alert tones – Within this section you can set up different alert tones for specific situations. Normal – It sets alert tone which is used at all cases except following two. Queued – It sets alert tone, which is used by routing to the station with active queue.
Routing objects 7.2 Figure 2 View of configuration menu of routing object Bundle – tab Advance. 7.2.2 DISA DISA The routing object DISA (Direct Inward System Access) is used for automatic call acceptance. Subsequent you can dial another digits via DTMF and route call through the PbX to the requested destination. In conjunction with suitable routers, you can create complex IVR structure.
Routing objects 7.2 Figure 1 View of the possible configuration of the DISA routing object. This menu consists of many parameters with following meaning: Tone – With this option you can choose suitable progress tone from the list. Your own progress tones and messages you can add in the menu 6.6 Progress tones. Strategy – This option sets way of using of the whole routing object DISA. You can choose between two strategies: Immediate – This strategy represents common concept of the DISA.
Routing objects 7.2 7.2.3 Ring groups Ring group The ring group is a routing object which is used in the cases when you want to route incoming call to more destinations at the same time. The ring groups are also used as user groups within which users can take over their calls. It is useful for situations when one of the users isn't present and his (or hers) phone is ringing. For this purpose are intended services Take over from group and Take over from my group.
Routing objects 7.2 Figure 1 View of configuration menu of routing object Ring group – tab Basic. Advanced settings Force CLIP – It is used as quick identification tab for incoming way differentiation. Calling party identification is changed after passing this object. You have to set up Scheme (Number or URI), Subtype (Unknown, Internal, Local, National, International) and Number/URI (specific number or address). In call history at the called user is presented original identification of the calling user.
Routing objects 7.2 Figure 2 View of configuration menu of routing object Bundle – tab Advanced. 7.2.4 Ring tabs Ring tab The ring tab in a routing object, which is used in the cases when you want to route incoming call subsequently to more destinations. It combines advantages of bundle and ring group. Routing of the incoming call is handled according to the preset rules, which are always executed from the beginning.
Routing objects 7.2 of the parameter Queue at the final destination. Route – Selected alert tone is used only in the case that on the final destination is enabled parameter Queue. No port station – It sets alert tone, which is used by routing to the user with station without carrier (external station). That user have to have assigned external station and at least one other internal station (which is active). Otherwise you will hear another alert tone.
Routing objects 7.2 End of routing With these commands you can terminate routing to that objects, which you have started to alert within this ring tab via commands Route or Route with queue. Don't route – With this command you can terminate routing to the specific object. You can terminate only routing to the objects, which you started to alert within this ring tab. For example you aren't able to terminate routing to the specific station of the user if you have started to route the call to the user.
Routing objects 7.2 Figure 2 View of configuration menu of routing object Ring tabs – tab Advanced. 7.2.5 Modems Modem connection Connection to the PbX via modem is used for remote access in the cases when you can't use TCP/IP connection. Modem you can also use for remote access to the database and for receiving actual system traces via application TraceView. This type of access is limited by its data rate and it isn't recommended in locations, where you can use TCP/IP access.
Routing objects 7.2 Modem setting Trace send enabled – With this option you can enable trace sending for application TraceView via modem. If this option isn't checked, application is connected, but no information is send to the remote user. In this mode you can view only database. Peer authorize required – With this option you can enable demand for connection dialogue box for access via modem. If this option isn't checked, is connection established without requirement of login and password.
Routing objects 7.2 stations, users, carriers, modems, DISAs and services. You have to think of the fact, that chaining will be probably finished if you use some object, which has no opportunity to the return to the set. It is recommended to add these objects to the end of the structure. The option Default is used in this column for return to the foregoing set (if you use set within the set). Id – With this column you can set specific object of the selected type.
Routing objects 7.2 Figure 1 View of configuration menu of routing object Audio I/O. Example 1 – Broadcast If you want to use audio port as source for your broadcast, you have to set selected port in menu Boards as Output and then assign it to the selected routing object Audio I/O. Using of broadcast is activated by incoming call to this routing object. If we want to play some announcement (e.g. We are beginning... 5, 4, 3, 2, 1, on air...), we can choose it within parameter Tone.
Routing objects 7.2 Binary ports Audio/IO/Relay board can have four or eight binary ports. Each port can be used in mode Output (switch), Input (detector) or in bidirectional mode (switch and detector). Function of each port depends also on hardware setting of jumpers. Individual useable modes are described in hardware manual in the chapter 2.14 Audio/IO/Relay board. Binary ports can't be used as other virtual ports of the PbX. You have to assign them to concrete routing object Binary I/O.
Routing objects 7.2 None – Routing object doesn't react to timeout expiration or end of the played tone. Hang up – After timeout expiration or playing whole tone is call hung up in routing object with cause no. 16 – normal call clearing. Call destination – After timeout expiration or playing whole tone is call routed according to configuration of section Destination. Play whole tone – With this option you can enable playing whole tone independently of preset timeout.
Routing objects 7.2 you see here Unknown, then is assigned binary port probably configured as Output or is this port or whole board out of order. Tone connected – It sets announcement played to the calling user when detector is in state active. Tone disconnected – It sets announcement played to the calling user when detector is in state deactive. Tone events enabled – It sets announcement played to the calling user in the case of checked option Send events.
Routing objects 7.2 sent and button Enable isn't activ. If sending is stopped, messages aren't sent and button Enable is ready for use. State detector – active – With this option you can enable message about active state of detector. Within this section you can define text of sent message. Optionally you can also stop another sending after this message via checkbox Stop sending when message was sent. State detector – inactive – With this option you can enable message about deactive state of detector.
Routing objects 7.2 Destination after timeout – This destination is used in the case of expiration of Ring detection timeout for another call routing. In the case of SMS Callback this destination isn't used. SMS form Incoming SMS for Callback has to be routed into text router where it can be routed to the selected object Callback. Used SMS has to be in following form: Called number,Delay,Calling number Called number – This parameter is mandatory.
Identification tab 7.3 7.3 Identification tab What is it Identification tab? The identification tabs are used for changing calling party number of the stations. You can create and modify them in the menu Routing – Identification tabs. If you want to use specific identification tab you have to assign it to the virtual port or virtual port type. Menu for set up consists of two windows. On the left side of the menu is displayed list of the created identification tabs.
Identification tab 7.3 Configuration of Identification tab If you select some identification tab on the left side of the menu, you can setup it on the right side. Window for configuration can be logically divided into four parts – Calling party determination, New identification determination, Advanced settings and Default destination. Calling party determination Calling party determination is performed at the beginning of each identification table row.
Autoclip routers 7.4 Default identification The last field is on the bottom of the menu and it is used for identification setting of all calling parties, which weren't found within the tab. Function of parameters within this part is same as by parameters from yellow highlighted part, which were described before. 7.
Autoclip routers 7.4 to the same number. This strategy refers to records storing and also to using of them. You can choose between three strategies: All – All records are saved to the database. If incoming call corresponds to more autoclip router records, all matching users are alerted at the same time. In sequence – If incoming call corresponds to more autoclip router records, all matching users are alerted subsequently one by one (from the newest record).
Autoclip routers 7.4 Port – This column shows port, which was used for routing of the outgoing call, which created this autoclip router record. It is used in the case of checked option Check port. Final destination – This column shows calling party, which created this autoclip router record. In the case of PbX user is here displayed his name and in the case of external user is here only his identification (CLI). To these destinations is also routed incoming call or SMS message.
Users and Groups 8.1 8 Users 8.1 Users and Groups Users creation Settings for users you can find in the menu Routing – Users and Groups. in this menu you can manage also groups and stations. On the left side of this menu is displayed list of created groups, subgroups, users and stations. Figure 1 Structure of PbX users from groups to the stations. Within context you can find following options: Add group – This option is used for basic groups adding.
Users and Groups 8.1 you will see dialog box from figure 2. Via this dialog box you can set another parameters and even create some stations which will be assigned to the new created user. Figure 2 View of the dialog box for creation of new user. In following part of this chapter are described individual tabs of the menu Users and Groups: Basic On the right side of this tab is displayed only name of selected group or subgroup.
Users and Groups 8.1 Properties The tab Properties consists of many tabs, which are described in the separate chapter. This tab is exceptional, because almost for all its parameters you can use hierarchical structure. This structure and description of all parameters is situated in the chapter 9.1 Setting up the properties tab. Profiles For easier handling with user settings were created user profiles, which provide easy changes of a large number of parameters in one step.
Users and Groups 8.1 c) Properties The tab Properties consists of many tabs, which are described in the separate chapter. This tab is exceptional, because almost for all its parameters you can use hierarchical structure. This structure and description of all parameters is situated in the chapter 9.1 Setting up the properties tab.
Users and Groups 8.1 Scheme – This column sets scheme of the user identification. You can choose between "Number" and "URI". Subtype – This column sets subtype of the user identification. You can choose between Unknown, Internal, Local, National and International. Number – Into this column you have to fill in user Number or URI according to column Scheme. Ring pattern – For each record of user phone book you can choose different ring tone.
Users and Groups 8.1 CFU – Call Forwarding Unconditional – This parameter sets unconditional forwarding to the voicemail. It means that all incoming calls will be forwarded directly to the voicemail if no profile is active (and no exceptions are set). CFEC – Call Forwarding on Error Cause – This parameter sets forwarding to the voicemail in the case of busy user or in the case of other error cause detection (e.g. call rejection).
User rights 8.2 Repeat try count – This parameter sets maximum count of sending attempts at fail for outgoing SMS messages. Preset value is 4000. Autoclip parameters for messages – With this parameter you can assign set of AutoClip parameters. Into the AutoClip router are then all records about outgoing SMS messages for this user stored with these parameters. Assistant This tab can be used only on user level.
User rights 8.2 Figure 1 View of logins divided according to the groups. Basic After selecting concrete user is on the right side of tab Basic displayed list of users who belong to the same group as selected user. For these users you can see its login types and there are also displayed all set rights. This view is useful when you need to set similar rights within group of users. Table of rights is divided into sections with following meaning.
Station types 8.3 NsAdmin This section isn't implemented yet. Assistant After selecting concrete user is on the right side of tab Assistant displayed list of users who belong to the same group as selected user. For these users you can see its login types and there are also displayed all set rights for application Assistant. This view is useful when you need to set similar rights within group of users. Table of rights is divided into sections with following meaning.
Stations 8.4 this station can't be created and you will be warn. Each station has to be assigned to specific user. This step is also done within this dialogue box. Futhermore you have to fill in the station number and if you are creating external station, you have to fill in the routing number too (number for call routing within other networks). In the case of external station you can also check the option Resend SMS, which enables to resend all incoming SMS messages to the external station.
Stations 8.4 Active – With this option you can activate or deactivate selected station. Deactivated station is unreachable for other stations (incoming call to this station is rejected), but station itself is able to establish outgoing calls. SIP – With this option you can mark SIP station among another stations. SIP station is then highlighted by green colour. Stations which are SIP and external together are red highlighted.
Phone books 8.5 assign selected station to one of the created terminals. c) SIP terminals are identified by their SIP URI. Terminals you can create at the SIP proxy carriers. With option Terminal you can easily assign selected station to one of the created terminals. Properties The tab Properties consists of many tabs, which are described in the separate chapter. This tab is exceptional, because almost for all its parameters you can use hierarchical structure.
Phone books 8.5 The phone book tab in this menu consists of individual records, which are devided into six columns with following meaning: Nickname – In this column you can set name which will be used for easier searching within the phone book. Name – It shows name of the station for which was this record created. This name will be displayed on your phone. Scheme – In this column you can set if entered string have to represent number or URI. Subtype – In this column you can set subtype of entered number.
Phone books 8.5 Name – It shows name of the station for which was this record created. This name will be displayed on your phone. Scheme – In this column you can set if entered string have to represent number or URI. Subtype – In this column you can set subtype of entered number. You can choose between subtypes Unknown, Internal, Local, National and International. It is active only if you use for this record scheme number. Number/URI – This column represents phone numbers (event.
Phone books 8.5 8.5.4 Common phone books Common phone books you can create in the menu Phone books – Common. There you can create "unlimited" number of phone books, which are then assigned to the selected groups of users. The context menu invoked on the right side of this menu offers following options: Within context menu on the right side of the menu you can use following optionst: Add – It is used for adding of another row of the selected phone book.
Setting the properties tab 9.1 9 Setting up the Properties Tab 9.1 Setting the properties tab Fall-down hierarchy All parameters on the tab Properties are used according to the fall-down hierarchy of the PbX. It means, that if you set some parameter on one specific level it isn't certain that it will be used. Each level of this fall-down hierarchy has preset priority. Following figure defines all levels of used falldown hierarchy. Higher situated levels have also higher priority.
Setting the properties tab 9.1 can find on all levels. a) Basic Parameters of this tab are mostly divided into sections according to their function: No answer timeout [ms] – This parameter sets maximum time of alerting of the called phone. After expiration of this timeout is call establishment terminated with cause "user not responding" and calling user hears congestion tone. Default value of this parameter is 180s. Next call – With this parameter you can set up caution about incoming queued call.
Setting the properties tab 9.1 delay). Identification parameters Incoming hold CLIP – This parameter is used for called party number forwarding to the called user in the case of call transfer made by station where is this parameter enabled. It means, that if it is set to "YES", transferred call will be identified by the calling number of transferred user (A) and not with the called party number of the user who transferred (B). Default value of this parameter is "NO".
Setting the properties tab 9.1 is used for call holding. If you use this pattern when the call is in hold, you can reconnect that call. If one call is active and second in hold, you can switch between calls using this pattern. DISCONNECT pattern – With this parameter you can set up characters of the DISCONNECT pattern. This pattern you can use when you have one call active and one in hold. Using this pattern you will terminate active call and call in hold will be reconnected.
Setting the properties tab 9.1 column Tone you can select required progress tone from another diaplayed list. In the case of dial tones is possible that for incoming call are satisfied more rules. Then is always used that rule, which is higher in configuration. On the right side are two buttons (arrows) which can be used for changing priority of already created rows. g) Ring patterns On this tab you can assign ring tones according to the calling party.
Setting the properties tab 9.1 extenders with ninety keys. In the case of terminals Entry, Economy, Basic, Standard and Advanced you can set up following parameters: 1) Keypad setting Keys are programmed via dialog box which you can invoke by clicking on the selected key. Within the dialog box you have to select function and legend of the key. You can choose between following functions: NONE – With this function you can clear function of selected key on the specific level of fall-down hierarchy.
Setting the properties tab 9.1 Keys volume – With this parameter you can set up loudness of the key hit in the handset or handsfree. Parameter can be set within the range of 0 to 15. Ring volume – With this parameter you can set up loudness of the ring tone. Parameter can be set within the range of 0 to 8. Handsfree volume – With this parameter you can set up loudness of the handsfree. Parameter can be set within the range of 0 to 15.
Setting the properties tab 9.1 seven levels from "extremely fast" to "extremely slow". i) AoC On the tab AoC you can set up warranted count of displayed call records on the system phone. You can set up independently count of records for Missed, Received and Dialed calls. Default count of records is set to twenty. This limitation you can notice only in the PbX with great number of users (carried calls). If the PbX has capacity to keep in store more records, all are displayed at any time.
Setting the properties tab 9.1 l) Customer On the tab Customer you can find parameters for functions which are implemented for specific customers and its meaning will be introduced only marginally. This tab is divided into the three sections. Within the first one you can define supported method of called party number sending. This number is used for billing. Within other sections you have to set up specific called party identification, which will be transmitted via DSS1 message.
Setting the properties tab 9.1 2N TELEKOMUNIKACE a.s. Modřanská 621, 143 01 Praha 4 Tel.: +420 261 301 111, Fax: +420 261 301 999 E-mail: sales@2n.cz Web: www.2n.cz v2.5.
